Many electronic instruments involve the technology of sampling, where sounds are digitally recorded and played back at different pitches. Sampling has the advantage of highly accurate and realistic sound. Once the sounds are recorded, however, it is difficult to change them in any significant way.
Traditionally, dynamic digital filtering for audio signals has included controlling a frequency response of an applied filter. As has been previously disclosed in U.S. Pat. No. 5,170,639, entitled “Dynamic Digital IIR Audio Filter and Method Which Provides Dynamic Digital Filtering for Audio Signals” issued Dec. 8, 1992 by the same inventor, incorporated by reference herein, one technique includes reducing the required multiplier coefficient size and allowing the fixing of a direct current gain of an infinite impulse response filter. Furthermore, music synthesizers released by E-mu Systems, Inc., including MORPHEUS and ULTRAPROTEUS, provided the ability to smoothly change a filter function over time. However, conventional systems were unable to provide an unconditionally stable filter structure or morph in multiple axes in real-time. Furthermore, traditionally the pole and zero frequency and amplitude have not been completely independent, which may cause undesirable interaction. While the basis of morphing a digital audio filter has been previously described, there are several improvements beyond the traditional music synthesizers, which are described herein. As such, a long felt need exists to provide improved dynamic digital filtering for audio signals to address one or more of these limitations.